* QCELP decoder
* Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file qcelpdec.c
+ * @file
* QCELP decoder
* @author Reynaldo H. Verdejo Pinochet
- * @remark FFmpeg merging spearheaded by Kenan Gillet
+ * @remark Libav merging spearheaded by Kenan Gillet
+ * @remark Development mentored by Benjamin Larson
*/
#include <stddef.h>
#include "avcodec.h"
-#include "bitstream.h"
+#include "internal.h"
+#include "get_bits.h"
#include "qcelpdata.h"
#include "celp_math.h"
#include "celp_filters.h"
+#include "acelp_filters.h"
+#include "acelp_vectors.h"
+#include "lsp.h"
#undef NDEBUG
#include <assert.h>
uint8_t erasure_count;
uint8_t octave_count; /*!< count the consecutive RATE_OCTAVE frames */
float prev_lspf[10];
- float predictor_lspf[10];/*!< LSP predictor, only use for RATE_OCTAVE and I_F_Q */
+ float predictor_lspf[10];/*!< LSP predictor for RATE_OCTAVE and I_F_Q */
float pitch_synthesis_filter_mem[303];
float pitch_pre_filter_mem[303];
float rnd_fir_filter_mem[180];
float pitch_gain[4];
uint8_t pitch_lag[4];
uint16_t first16bits;
-} QCELPContext;
+ uint8_t warned_buf_mismatch_bitrate;
-/**
- * Reconstructs LPC coefficients from the line spectral pair frequencies.
- *
- * TIA/EIA/IS-733 2.4.3.3.5
- */
-void ff_qcelp_lspf2lpc(const float *lspf, float *lpc);
-
-static void weighted_vector_sumf(float *out, const float *in_a,
- const float *in_b, float weight_coeff_a,
- float weight_coeff_b, int length)
-{
- int i;
-
- for(i=0; i<length; i++)
- out[i] = weight_coeff_a * in_a[i]
- + weight_coeff_b * in_b[i];
-}
+ /* postfilter */
+ float postfilter_synth_mem[10];
+ float postfilter_agc_mem;
+ float postfilter_tilt_mem;
+} QCELPContext;
/**
* Initialize the speech codec according to the specification.
QCELPContext *q = avctx->priv_data;
int i;
- avctx->sample_fmt = SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
for(i=0; i<10; i++)
q->prev_lspf[i] = (i+1)/11.;
}
/**
- * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
- * transmission codes of any bitrate and checks for badly received packets.
+ * Decode the 10 quantized LSP frequencies from the LSPV/LSP
+ * transmission codes of any bitrate and check for badly received packets.
*
* @param q the context
* @param lspf line spectral pair frequencies
lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
// Low-pass filter the LSP frequencies.
- weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
+ ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
}else
{
q->octave_count = 0;
}
/**
- * Converts codebook transmission codes to GAIN and INDEX.
+ * Convert codebook transmission codes to GAIN and INDEX.
*
* @param q the context
* @param gain array holding the decoded gain
gain[2] = gain[1];
gain[1] = 0.6*gain[0] + 0.4*gain[1];
}
- }else
+ }else if (q->bitrate != SILENCE)
{
if(q->bitrate == RATE_OCTAVE)
{
}
/**
- * Computes the scaled codebook vector Cdn From INDEX and GAIN
+ * Compute the scaled codebook vector Cdn From INDEX and GAIN
* for all rates.
*
* The specification lacks some information here.
* @param gain array holding the 4 pitch subframe gain values
* @param cdn_vector array for the generated scaled codebook vector
*/
-static void compute_svector(const QCELPContext *q, const float *gain,
+static void compute_svector(QCELPContext *q, const float *gain,
float *cdn_vector)
{
int i, j, k;
*cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
}
break;
+ case SILENCE:
+ memset(cdn_vector, 0, 160 * sizeof(float));
+ break;
}
}
* @param v_in gain-controlled vector
* @param v_ref vector to control gain of
*
- * FIXME: If v_ref is a zero vector, it energy is zero
- * and the behavior of the gain control is
- * undefined in the specs.
- *
- * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
+ * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
*/
static void apply_gain_ctrl(float *v_out, const float *v_ref,
const float *v_in)
{
- int i, j, len;
- float scalefactor;
+ int i;
- for(i=0, j=0; i<4; i++)
- {
- scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
- if(scalefactor)
- scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40)
- / scalefactor);
- else
- av_log_missing_feature(NULL, "Zero energy for gain control", 1);
- for(len=j+40; j<len; j++)
- v_out[j] = scalefactor * v_in[j];
- }
+ for (i = 0; i < 160; i += 40)
+ ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
+ ff_dot_productf(v_ref + i,
+ v_ref + i, 40),
+ 40);
}
/**
/**
* Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
- * TIA/EIA/IS-733 2.4.5.2
+ * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
*
* @param q the context
* @param cdn_vector the scaled codebook vector
const float *v_synthesis_filtered, *v_pre_filtered;
if(q->bitrate >= RATE_HALF ||
+ q->bitrate == SILENCE ||
(q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
{
}
}else
{
- float max_pitch_gain = q->erasure_count < 3 ? 0.9 - 0.3 * (q->erasure_count - 1) : 0.0;
+ float max_pitch_gain;
+
+ if (q->bitrate == I_F_Q)
+ {
+ if (q->erasure_count < 3)
+ max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
+ else
+ max_pitch_gain = 0.0;
+ }else
+ {
+ assert(q->bitrate == SILENCE);
+ max_pitch_gain = 1.0;
+ }
for(i=0; i<4; i++)
q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
}
/**
- * Interpolates LSP frequencies and computes LPC coefficients
+ * Reconstruct LPC coefficients from the line spectral pair frequencies
+ * and perform bandwidth expansion.
+ *
+ * @param lspf line spectral pair frequencies
+ * @param lpc linear predictive coding coefficients
+ *
+ * @note: bandwidth_expansion_coeff could be precalculated into a table
+ * but it seems to be slower on x86
+ *
+ * TIA/EIA/IS-733 2.4.3.3.5
+ */
+static void lspf2lpc(const float *lspf, float *lpc)
+{
+ double lsp[10];
+ double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
+ int i;
+
+ for (i=0; i<10; i++)
+ lsp[i] = cos(M_PI * lspf[i]);
+
+ ff_acelp_lspd2lpc(lsp, lpc, 5);
+
+ for (i=0; i<10; i++)
+ {
+ lpc[i] *= bandwidth_expansion_coeff;
+ bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
+ }
+}
+
+/**
+ * Interpolate LSP frequencies and compute LPC coefficients
* for a given bitrate & pitch subframe.
*
- * TIA/EIA/IS-733 2.4.3.3.4
+ * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
*
* @param q the context
* @param curr_lspf LSP frequencies vector of the current frame
* @param lpc float vector for the resulting LPC
* @param subframe_num frame number in decoded stream
*/
-void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
- const int subframe_num)
+static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
+ float *lpc, const int subframe_num)
{
float interpolated_lspf[10];
float weight;
if(weight != 1.0)
{
- weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
- weight, 1.0 - weight, 10);
- ff_qcelp_lspf2lpc(interpolated_lspf, lpc);
+ ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
+ weight, 1.0 - weight, 10);
+ lspf2lpc(interpolated_lspf, lpc);
}else if(q->bitrate >= RATE_QUARTER ||
(q->bitrate == I_F_Q && !subframe_num))
- ff_qcelp_lspf2lpc(curr_lspf, lpc);
+ lspf2lpc(curr_lspf, lpc);
+ else if(q->bitrate == SILENCE && !subframe_num)
+ lspf2lpc(q->prev_lspf, lpc);
}
-static int buf_size2bitrate(const int buf_size)
+static qcelp_packet_rate buf_size2bitrate(const int buf_size)
{
switch(buf_size)
{
case 1: return SILENCE;
}
- return -1;
+ return I_F_Q;
}
/**
*
* TIA/EIA/IS-733 2.4.8.7.1
*/
-static int determine_bitrate(AVCodecContext *avctx, const int buf_size,
- uint8_t **buf)
+static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size,
+ const uint8_t **buf)
{
qcelp_packet_rate bitrate;
{
if(bitrate > **buf)
{
+ QCELPContext *q = avctx->priv_data;
+ if (!q->warned_buf_mismatch_bitrate)
+ {
av_log(avctx, AV_LOG_WARNING,
"Claimed bitrate and buffer size mismatch.\n");
+ q->warned_buf_mismatch_bitrate = 1;
+ }
bitrate = **buf;
}else if(bitrate < **buf)
{
if(bitrate == SILENCE)
{
- // FIXME: the decoder should not handle SILENCE frames as I_F_Q frames
- av_log_missing_feature(avctx, "Blank frame", 1);
- bitrate = I_F_Q;
+ //FIXME: Remove experimental warning when tested with samples.
+ av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
}
return bitrate;
}
message);
}
+static void postfilter(QCELPContext *q, float *samples, float *lpc)
+{
+ static const float pow_0_775[10] = {
+ 0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
+ 0.216676, 0.167924, 0.130141, 0.100859, 0.078166
+ }, pow_0_625[10] = {
+ 0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
+ 0.059605, 0.037253, 0.023283, 0.014552, 0.009095
+ };
+ float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
+ int n;
+
+ for (n = 0; n < 10; n++) {
+ lpc_s[n] = lpc[n] * pow_0_625[n];
+ lpc_p[n] = lpc[n] * pow_0_775[n];
+ }
+
+ ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
+ q->formant_mem + 10, 160, 10);
+ memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
+ ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
+ memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
+
+ ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
+
+ ff_adaptive_gain_control(samples, pole_out + 10,
+ ff_dot_productf(q->formant_mem + 10, q->formant_mem + 10, 160),
+ 160, 0.9375, &q->postfilter_agc_mem);
+}
+
static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
- uint8_t *buf, const int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
QCELPContext *q = avctx->priv_data;
float *outbuffer = data;
int i;
10);
formant_mem += 40;
}
- memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
- // FIXME: postfilter and final gain control should be here.
- // TIA/EIA/IS-733 2.4.8.6
+ // postfilter, as per TIA/EIA/IS-733 2.4.8.6
+ postfilter(q, outbuffer, lpc);
- formant_mem = q->formant_mem + 10;
- for(i=0; i<160; i++)
- *outbuffer++ = av_clipf(*formant_mem++, QCELP_CLIP_LOWER_BOUND,
- QCELP_CLIP_UPPER_BOUND);
+ memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
q->prev_bitrate = q->bitrate;
return *data_size;
}
-AVCodec qcelp_decoder =
+AVCodec ff_qcelp_decoder =
{
.name = "qcelp",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_QCELP,
.init = qcelp_decode_init,
.decode = qcelp_decode_frame,
projects / ffmpeg / blobdiff